In some tonelists, as used in Playtones or indications.conf, I've seen a
notation to set levels, for instance [EMAIL PROTECTED] The -10 doesn't
seem to do anything. Is there a patch that will enable setting levels in
a tonelist?
___
Asterisk-Users
Guys.
I just installed spandsp and configured asterisk for receiving faxes but my
first test came out wrong, all 3 pages of the fax were cut off.
My test was using one of my ATAs connected to a modem on a PC and dialed
into the asterisk using ALAW. Im using spandsp pre15 and asterisk cvs head.
Hi Andrew,
I've experiened the noisy line from both my POTS lines (Digit Networks X101P
cards -- 3 of them to be exact) as well as my VOIP provider (Broadvoice)...
The problem has also occured in both directions (I.E I originate the call,
or someone has called me, again on 2 POTs lines (3rd is
Have you had some experience with GS ATA 286? I have 2 analog phones
connected and using the latest firmware on the ATAs and from time to time,
while in a call, the line just gets filled with line noise and you have to
hit flash and then flash again to retake the call and the line noise is
cool. see, no need to fight anyone. you people are crazy.
luf...
On Wed, 20 Apr 2005, trixter http://www.0xdecafbad.com wrote:
you did a great parody of him completly ignoring what I was saying and
going off on something unrelated to what I say just to get MS bashing
in. Gotta love people who
spandsp does fax. not fax over ip but just fax.
On 4/21/05, Anton Krall [EMAIL PROTECTED] wrote:
Guys.
I just installed spandsp and configured asterisk for receiving faxes but my
first test came out wrong, all 3 pages of the fax were cut off.
My test was using one of my ATAs connected to
stop wasting my bandwidth plz
On Wed, 20 Apr 2005, trixter http://www.0xdecafbad.com wrote:
On Wed, 2005-04-20 at 22:14 -0500, Dan Perik wrote:
Michael D Schelin wrote:
Ok you guys enough. The debate will go on forever.
Agreed! At the risk of wasting bandwidth myself
Please, guys stop
All depends what you call a large installation
On 4/21/05, Callum McGillivray [EMAIL PROTECTED] wrote:
We are currently planning a large solution for a similar type of
scenario, ours is an implementation for a hotel with 600+ rooms. The
core differences being that we will have a greater flow
On Thu, 2005-04-21 at 09:06 +0200, Michael Bielicki wrote:
spandsp does fax. not fax over ip but just fax.
Wonder if anyone would (or has) write a fax gateway app that would read
it off the PRI or whatever then store and forward. Would enable
'internet faxing' without the requirement for T.38
I (601) call one of my users (8862), after one minute I try to call him
again and get Unable to create channel of type 'SIP'
sip show peers does not list him.
I cannot figure out why this happens and more important how I can fix it.
-- Executing Dial(SIP/601-0f22, SIP/8862|60|tr) in new
I was doing fax over IP, a PC was connected to an ATA which used ALAWY and
dialed into the fax extension, both the PC and asterisk are local here.
Problem is that asterisk and the PC reported the fax came out right but when
opening the tiff file, all 3 pages were cut off at the beginging.
On Thursday 21 April 2005 09:38, Ronald Wiplinger wrote:
I (601) call one of my users (8862), after one minute I try to call him
again and get Unable to create channel of type 'SIP'
sip show peers does not list him.
I cannot figure out why this happens and more important how I can fix it.
On 4/19/05, Charles Wang [EMAIL PROTECTED] wrote:
Dear ALL:
My scenario is:
SIP UA == SIP Proxy == Asterisk == CISCO 5300 trunk == PSTN
I make a call from SIP UA to a PSTN phone number, and SIP UA hangup first.
My Asterisk can receieve a BYE message, so this connection will be hangup.
Hello all,
I have got a new project with asterisk, That it is to replace the
existing PBX on a company.
This company, has one remote officce. From one officce you can dial an
extension and talk with the other end.
There are a tunnel between the offices, using cisco routers for data
and
List Members,
Hi !
..
The goal of our new design is to offload the DSP to the
Asterisk slave
servers, then route the calls via IAX2 trunks to the Asterisk master
server. The Asterisk master server will provide us with a
centralized
point for queuing, digital recording, and music on
I'm trying to use some VoIP phones behind a Linksys WRT54G router but can't
get them to register.
I am doing this now.
Am I missing some trick to get Linksys to cooperate with my asterisk setup?
Are you forwarding ports on the WRT54g to the phone?
On the phone I'm using BT100 .. and I
I want to set up a * box that connects to my telephone line so I can
make calls using VOIP. Do I have to sign up for a service to do this
Yes, read the first doc below.
can I buy a card that goes into my * box that allows me to use VOIP over
my telephone line. If so, what card do I need? I
Hello all,
I have got a new project with asterisk, That it is to replace the
existing PBX on a company.
This company, has one remote officce. From one officce you can dial an
extension and talk with the other end.
There are a tunnel between the offices, using cisco routers for data
and
Hello
List,
I´m using Asterisk
1.0.7 with suse 9.2 most works fine but sounds from asterisk are choppy. I found
that is it necessary to use the kernelmodul ztdummy for timing if no zaptel
device is present.
Ok i can build the
kernelmodule but can not loading it.
dmesg
say´s
module
Hello List,
I have an * for a small office using SIP Softphones. We use for attended
Transfer call parking and so on ..
Now i read in the ip-phone forum that a new option for attended transfer is
given Where can i get info´s about this feature.
Greets harry
Has anybody success with speed dialing?
If so, I am sure you can help me to get into this club.
tgj wrote:
Hi Ronald,
It seems like you need to put in default as your context. However I think
your problem was that you put the number in CallerID column and The CallerID
in the Name column. I was
Hi all
Could someone please care to share an example of the Dial W option
usage. I cannot seem to find any reference to it usage. I know you use
*1 in features.conf to start the monitor, but from there I am lost.
Master
___
Asterisk-Users mailing list
Hi, please help on setting up DID with similar extension number.
Thanks.
-Original Message-
From: Nathaniel Angelo A. Torres (247talk) [mailto:[EMAIL PROTECTED]
Sent: Wednesday, April 20, 2005 10:08 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] DID ~ Extension
Hi, any lead
Thanks
Which config file does this need adding to then? I fail to see any
mention of Dialpan
Also will doing a Re-Read Configs from the web config page suffice
for restarting and loading new config of Asterisks?
On 4/20/05, Tim Thompson [EMAIL PROTECTED] wrote:
You will need to make sure that
Hi Mark
You need to close the ( :
exten = s,2,GotoIf($[${CALLERIDNUM} = xx]?4)
Regards,
Fred
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Mark
Halverson
Envoyé : mercredi 20 avril 2005 21:59
À : 'Asterisk Users Mailing List - Non-Commercial
Hello
I have capi 0.3.5 with fritz card and:
When i try to dialout i have this message in from capi debug :
CAPI Debugging Enabled
CLI -- Executing AnswerSIP/478-3f3f in new stack
--
Hello, I am having some problems using Asterisk together with ALSA and
OSS with two different motherboards with VIA686 soundcards and Linux
kernel 2.6.10.
The ALSA drivers work but have a latency of around 0.5seconds and the
OSS-driver only works sometimes on one of the computers but never on the
This is my sip trunk configuration
canreinvite=yes
context=from-internal
disallow=g729
host=192.168.1.138
mask=255.255.255.255
qualify=yes
type=peer
maybe it could be the context!
HTH
Edgar
On Thu, 2005-04-21 at 12:57 +0800, Dinesh wrote:
Great:)
Just one question, I am trying to get the
Since capi is not even really supported by the guy who wrote it I'd
suggest you get a zaphfc card, like the I-tec ISDN-128, which could
simplify your task a lot.
:)
cheers Micha
On 4/21/05, Pawe Staszewski [EMAIL PROTECTED] wrote:
Hello
I have capi 0.3.5 with fritz card and:
Are youy using some old GS firmware ? Works without problems here.
On 4/21/05, Wilson Pickett [EMAIL PROTECTED] wrote:
I'm trying to use some VoIP phones behind a Linksys WRT54G router but can't
get them to register.
I am doing this now.
Am I missing some trick to get Linksys to
I was thinking about the best way to hook up the second line in my
house to an * fxs port. Would I just wire the fxs to the incoming
side of a line at my demarc? Or should I splice it in after that?
I need to rewire the whole house anyways. What I had imagined was new
cat3 for the
I Don't know if this is a solution that is not better suited for your
Linux Distro, using vconfig and a 802.1q/1p aware NIC, on the * Server, to
create and prioitize the Ingress and Egress Voice VLAN traffic.
Would you not agree?
Sean
On Wed, 20 Apr 2005 21:14:17 +0200, Eugenio De Vena
Hi all
I try to implement an asterisk solution in my company.
I use Digium card and sip phone Budgetone 102.
When i call an external number, I hear myself during 3 or 4 sec and then it
diseapear progressivly.
Do you have any idea so as to solve my problem
Sorry for the english.
MDEL
France
Hi
How can I Deny certain extension within certian context from calling outside
calling.and let him to call another extension within his context
regards
_
Express yourself instantly with MSN Messenger! Download today it's FREE!
On April 21, 2005 02:28 am, Andre Normandin wrote:
I've experiened the noisy line from both my POTS lines (Digit Networks
X101P cards -- 3 of them to be exact) as well as my VOIP provider
(Broadvoice)...
The problem has also occured in both directions (I.E I originate the call,
or someone
Hi Rafa!
Hi Jairo,
Try with other values for the jitter in your Gateway
(H323).
One customer have a scenario like this:
Phone/Fax Gateways H323 -with 16/8/2/1 Port
FXS- --- GNUGK
--- Asterisk --- Zap (E1)
We have the same configuration, and I think that our
problem is the jitter in
On April 21, 2005 08:08 am, Ashraf Salah wrote:
How can I Deny certain extension within certian context from calling
outside calling.and let him to call another extension within his context
This is what contexts are all about.
If it's an extension off of a Zap card, but give them a specific
On April 21, 2005 08:07 am, Michaël Delvoye wrote:
I try to implement an asterisk solution in my company.
I use Digium card and sip phone Budgetone 102.
When i call an external number, I hear myself during 3 or 4 sec and then it
diseapear progressivly.
google or wiki around for
On Thursday 21 April 2005 13:07, Michaël Delvoye wrote:
Hi all
I try to implement an asterisk solution in my company.
I use Digium card and sip phone Budgetone 102.
When i call an external number, I hear myself during 3 or 4 sec and then it
diseapear progressivly.
Do you have any idea so as
That's a new one. Occasionally they show up dead, but usually if they work
the sound quality is excellent. I'll forward this on to Grandstream. In
the meantime, please post it to the newsgroup. -Mike
http://www.thevoipconnection.com/forums/index.php?board=4.0
Michael Crown
Managing Partner
The
I drop every 3-4 call with VoicePulse Connect.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Kennedy
Sent: Wednesday, April 20, 2005 6:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice
On Thu, 2005-04-21 at 12:09 +0200, Pawe Staszewski wrote:
Hello
I have capi 0.3.5 with fritz card and:
When i try to dialout i have this message in from capi debug :
CAPI Debugging Enabled
*CLI -- Executing Answer(SIP/478-3f3f, ) in new stack
-- Executing
SNIP
== DISCONNECT_IND PLCI=0x101 REASON=0x3481
== No one is available to answer at this time
How changing from CAPI to a zaphfc card will correct
this error I don't
know, and problably neither does the person who
suggested it.
REASON 0x3481 is Unallocated (unassigned) number. =
Wrong
they will all work, I use Mandriva, If you want something solid look
at CentOS, Debian is cool but the stable branch is out dated while the
development branch is rather stable. SUSE could work but I don't like
to pay for a distro right up front. I do suggest that you donate to
the distro that you
This all assumes that CF cards are more reliable than hard
drives (and power supplies).
That seems to be the commonly accepted fact. With no moving parts, purely
silicon, it should be as reliable as any electronics can be. Hard drives are
notoriously failure prone.
The best solution is a
Using most recent CVS-HEAD and my terminal keeps changing colors.
I'm using vt100 terminal emulation. How can I turn off asterisk's colors? Or
at least turn off the black background. My normal terminal is white
background, black font. But for some reason, asterisk is changing it to
white font,
Andrew Latham wrote:
they will all work, I use Mandriva, If you want something solid look at CentOS,
But, like all the later RH offshoots, there are problems with some
hardware. One annoying little problem is Kudzu killing a 3Com network
card after installation. This problem has been around
Nice setup. Please do feed the list with your findings.
I'm interested in a solution for digitally recording calls and I
noticed you have a Digital Recording Client. Would you mind elaborating
more on this? Is this another Asterisk box issuing the Monitor command?
If it is, does it mean that
In particular have a look at
http://www.voip-info.org/wiki-Asterisk+x100p+echotraining
Using this guide you can set * to send an impulse down the line so that
it trains much quicker and also manually adjust the tx and rx gain to
improve things furthur.
On Thu, 2005-04-21 at 13:19, Gavin Hamill
Hi
all,
Does anyone has any
experience with bristuff in Belgium... ???
I have an ISDN line
at home and I try to install asterisk (the bristuff version 1.0.6). It
works although I received a lot of messages from zaphfc telling me it didn't
receive the correct number of frames for both
If your used to RH keep using it. Since I am a person that has used RH for
many years I have gone with CentOS which is RHEL via GPL. It's great and
there yum servers are always up and running.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Shiflet
Hello
I live in poland and :
local numbers are: 752 7 digits
zone prefix: 32
country prefix: 48
And i must add that i am behind a local PBX Alcatel 4200E
Configured isdn port with msn 7523071
Why
trixter http://www.0xdecafbad.com; [EMAIL PROTECTED] wrote:
Wonder if anyone would (or has) write a fax gateway app that would read
it off the PRI or whatever then store and forward. Would enable
'internet faxing' without the requirement for T.38 or similar.
There is already a spec for this.
Hi,
I'm starting to play with Asterisk and the queues. Although I have
found many references to the queues.conf in the wiki, I haven't been
able to find answers to the following:
1) If I understand correctly, an agent can belong to more than one
queue. Does that mean that as soon as the agent
On Wed, Apr 20, 2005 at 10:26:33PM -0500, Paul Shiflet wrote:
I'm trying to find out what flavor of Linux people are choosing for their
asterisk boxes. I have been using RH, but i'd like to try some different
ones. It seems that RH is the common denominator in this rash of line
noise problems.
I got it to work. I didn't set the source ip address of my Asterisk server
in the sccp.conf file. They work like a charm now. Thanks for the offer
though.
Chuck
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andy Hamilton
Sent: Wednesday, April 20,
Hello,
Actualy I have got a PBX conected to a BRI port on a Cisco Router witch route the calls. The router is using h323 with the other end.
How could I replace the actual PBX and pass all the calls to the router, as now works.
I tryed connecting an asterisk CAPI port against the same BRI
I have two Asterisk boxes installed but am not sure how to setup the
configuration for what I want to do.
One box has two FXO cards in it that will connect two PSTN lines. I want to
have Asterisk transfer the incoming calls to the other box which has an FXS
card in it. That box should ring
When trying to send calls from our Asterisk PBX to our upstream
termination provider, I am getting
Got SIP response 503 Service Unavailable back from PROVIDER
We are sending the calls without registration, there is no username and
password. When we were using SER it would send them without a
As of December, hints did not work with a Realtime config in asterisk.
Here's a link to that discussion:
http://lists.digium.com/pipermail/asterisk-users/2004-December/077410.html.
My current configuration is Asterisk CVS-HEAD-04/15/05-12:52:11 with
Realtime connected on a mysql database, SIP
I am trying to get * to send a SIP notify to my SIP phone when an IAX2 channel
goes active.
* (1.0.5) is accepting the Subscribe and sending the notify just fine with the
SIP channels and Zap channels, but not this IAX2 channel.
I have this in my context as the hint:
exten = 200,hint,IAX2/[EMAIL
I have a customer with 2/3rds of a PRI turned up for his PM3, which handles
incoming calls for dialup customers. He's offered me the remainder of the PRI
at cost, assuming he can still get calls through to his PM3.
I know it's fairly easy to have incoming calls (with his DID) routed to his
PM3
On Wed, 20 Apr 2005, Noah Miller wrote:
Is this to disable the call waiting on Polycom phones? If so, I don't
think there wouldn't be a need to have it on the s extension. You'd
just need to have it on any extension that the Polycom phone would call
(other handsets, outside lines,
You don't have to use Sendmail (you can use Postfix too), but it has to
be something on the * server. It uses the local SMTP server to send the
email from the server. If it wasn't there, how would it get to the
other server?
list wrote:
All,
I'd like to use the Voicemail to Email feature of
Pawe Staszewski wrote:
Hello
I live in poland and :)
local numbers are: 752 (7 digits)
zone prefix: 32
country prefix: 48
And i must add that i am behind a local PBX (Alcatel 4200E)
Configured isdn port with msn 7523071
Why dial in is working but dial-out not ... ??
maybe your local PBX
On Thu, 2005-04-21 at 08:39 -0500, Matthew Boehm wrote:
Using most recent CVS-HEAD and my terminal keeps changing colors.
I'm using vt100 terminal emulation. How can I turn off asterisk's colors? Or
at least turn off the black background. My normal terminal is white
background, black font.
On 21/04/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
list wrote:
All,
I'd like to use the Voicemail to Email feature of asterisk, but I dont
want to use sendmail. We have a seperate email server that we would
like to use for this feature. How do and where do I specify this?
You
Hello
i try with 0 and
-- Executing AnswerSIP/478-c9a2 in new stack
-- Executing DialSIP/478-c9a2 CAPI/7523071:07522333 in new stack
-- data = ""
-- capi request omsn = 7523071
== found capi with omsn = 7523071
We used gentoo internally. I also have * running on CentOS, RHEL.
Best Regards,
==
David Choo
Systems Engineer
Business Technology Division
Engineered for Changing Businesses
Espore Corp Pte Ltd
68 Kallang Pudding Rd
#04-03 SYH Logistics Bldg
Singapore 349327
Tel:
On 4/21/05, Adam Robins [EMAIL PROTECTED] wrote:
I drop every 3-4 call with VoicePulse Connect.
My users are also reporting occasional dropped calls when dialing via
VoicePulse Connect.
But I love the ease of use and setup with their service.
-Trevor
Good day all
I want to put a asterisk server on a public ip and allow any,registered
sip and iax connection
What security risks are there and how can I secure my pabx
One thing I want to know is how do I make it that anyone can call a
extension at my box but not make a call out.
i.o.w how do I
On Thursday 21 April 2005 16:31, Peter Bowyer wrote:
Many similar systems (webmail, bulletin boards etc) are configurable
to use a local MUA (/sbin/sendmail etc) or talk SMTP directly to an
MTA, either locally or remote. Asterisk voicemail unfortunately is not
one of those systems (AFAICT) -
Hi All,
Manage to get it work. It is the voltage problem. The converter needs
9-20v adapter but I only provide 5v.
Stupid mistake.
It works well now.
Thanks,
Stephen
Peter Hoppe wrote:
Stephen,
It would be very kind if you provided the make and model of the
fxs-fxo converter. Also, what you
Has anyone managed to find what the maximum concurrent faxes, either
incoming or outgoing using SpanDSP? I realise that it depends on what
system it is running on, so some basic details of the system would
be appreciated.
B
___
Asterisk-Users mailing
Hi, Im receiving this error, please help me solve
this.
Apr 22 00:12:26 VERBOSE[3735]: == Parsing
'/etc/asterisk/phone.conf': Apr 22 00:12:26
VERBOSE[3735]: == Parsing '/etc/asterisk/phone.conf': Found
Apr 22 00:12:26 VERBOSE[3735]: ==
Registered channel type 'Phone' (Standard Linux
Daniel,
So far our digital recording client is a box on a diagram. I haven't
digitally recorded a single call through Asterisk yet, but this is a
learning process so I'll share what I know in hopes that I can help
people out and have any of my mistakes corrected.
As I understand it, digital
I just wanted to make everyone aware that I cross-posted my original
message to the Biz list. You may want to check out the responses there,
too.
It looks like the entire Asterisk slave server pool in my diagram
(http://home.comcast.net/~mroth01/LargeAsteriskSetup.gif) can be
replaced by a
Hi Cameron,
Yes I did looked at those pages, and tried to configure NAT and externip
and localnet and all that jazz, but still no audio. I'll definitely try
to look at those options agai, since the device on outside network can
communicate fine.
Sincerely,
--Andy
x6722
Outsourcing is akin to
Chris,
If you are looking to run your second line through *, you will need to
run the line from the demarc to an FXO port on your Asterisk machine,
and then run a line from an FXS card to the jack location where you
will be using an analog phone.
You cannot connect an incoming line from your
Angelo,
Looks like an error in zapata.conf (/etc/asterisk/zapata.conf). Without
seeing the file, I'd guess that you're missing a context somewhere.
It would be helpful if you posted the contents of the file to the list.
Matthew Roth
Hmmm... Think I would prefer something harder to get provisioned but
that doesn't drop calls.
Your users must be forgiving as hell... Mine would show up with
pitchforks and torches if calls dropped regularly.
They get twitchy if the calls just vary too much in quality... 8)
Cheers,
Wiley
well I thought that with diffserv it could be done, I will double check and
let you know,
thanks for you hint.
Eugenio
- Original Message -
From: SCollins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April
Terry:
You'll probably want to set up an IAX2 channel between them.
Here is a good place to start looking:
http://www.voip-info.org/wiki-Asterisk+IAX+Channels
Also, I think this list has had a fair share of IAX2 inquiries
recently. If you check the archives
I have two Asterisk boxes installed but am not sure how to setup the
configuration for what I want to do.
One box has two FXO cards in it that will connect two PSTN lines. I
want to
have Asterisk transfer the incoming calls to the other box which has
an FXS
card in it. That box should ring
Why wouldn't you just use tos=some value like 0xb8 or tos=lowdelay
depending upon the config file in question? Remember there is an overlap
between IP precedence bits and some DSCPs for backward compatability.
Honor that overlap and you can use DiffServ processing logic even if your
device can
Hi Fellow Asteriskians,
Currently I am having an Asterisk installation piggy-backed behind an
intelligent voice-switch that does the LCR, over E1's using Digium TE410p
The problem is that Asterisk always starts channel 1 if the channel is free,
and only switches to channel 2 if channel 1 is in
Just for grins A few thoughts.
Run Cat5 exclusively and just pull pairs for phone. Cheaper and better
solution that CAT3 and CAT5 mixed together.
It allows you to change the end points at will. Who knows if you may
want to change over to RJ45 ports and go total IP at some point.
You would
Try looking up smarthost for send mail. You can have a very basic
sendmail install and have it send mail via a smart host. That is what
I do it works well.
On 4/20/05, list [EMAIL PROTECTED] wrote:
All,
I'd like to use the Voicemail to Email feature of asterisk, but I dont want
to use
Hi,
Someone have more info about TE406P/TE411P ? (from VON europe mailing)
Stop by the Digium | Asterisk booth #1151 and check out the latest new
Digium products, including:
a.. Asterisk Business Edition
b.. DS3000P - Channelized DS3/T3/E3 voice and data PCI card
c.. TE406P TE411P -
I've used both Debian sarge and centos 4. If you compile * and zaptel
from source it's easier to do with centos due to how debian structures
it's kernel source and header files.
My personal favorite between the two is centos.
Chris
___
Hi, any idea what causes this and whats the solution.
then mv -f .deps/testcall.Tpo .deps/testcall.Po;
else rm -f .deps/testcall.Tpo; exit 1; fi
/bin/sh ./libtool --mode=link gcc -g -O2 -o testcall testcall.o
-lunicall -lpthread -laudiofile -ldl
gcc -g -O2 -o .libs/testcall
Paul, I prefer CentOS too as far as ease of * installation and ease of
updating. Being so similar to Redhat Enterprise Linux, this probably
isn't what you wanted to hear, but I can say that I have had no
problems with line noise!
Mojo
Paul Shiflet wrote:
I'm trying to find out what flavor of
Just to add some general wiring tips from my experience. (and yes you need
to go to an fxo port with the incoming telco line, and the fxs ports go to
your phones - you can't mix and match and parallel things up, the only
exception being more than one phone on an fxs port if the port can supply
Hi, here's the content of my Zapata.conf
[channels]
language=en
context=default
signalling=em_w
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
channel=1-15,17-30
I really don't know what to load into these values. I wanted to use E1R2 of
I totally concur. I switched from Broadvoice to VoicePulse because
users were complaining about call quality. Now, the quality is good --
when it doesn't drop altogether.
What could be worse than touting your new VoIP system to a client and
having it drop the call?
-Original
Pawe Staszewski wrote:
Hello
i try with 0 and
-- Executing Answer(SIP/478-c9a2, ) in new stack
-- Executing Dial(SIP/478-c9a2, CAPI/7523071:07522333) in new stack
-- data = 7523071:07522333
-- capi request omsn = 7523071
== found capi with omsn = 7523071
== CAPI Call
Here's my zaptel.conf
loadzone = us
defaultzone=us
span=1,1,0,cas,hdb3
#
cas=1-15:1101
cas=17-31:1101
and my Zapata.conf
[channels]
language=en
context=default
signalling=em_w
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
channel=1-15,17-30
Even if you choose not to use his other suggestions, I strongly agree with
Wiley's idea of using cat-5 instead of cat-3. The difference in cost is
minimal, and it will give you much more flexibility down the road. You
could even terminate the cat-5 with an RJ-45, and plug your RJ-11 phone cord
On Fri, 22 Apr 2005 01:26:45 +0800
Nathaniel Angelo A. Torres (247talk)
[EMAIL PROTECTED] wrote:
Hi, here's the content of my Zapata.conf
[channels]
language=en
context=default
signalling=em_w
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
We're using CVS-HEAD-04/21/05-11:34:04 and seem to be having problems with
Remote Agents receiving calls. When someone's in the queue, they get
dialed, but the AckCall '#' isn't being accepted and the call is simply
stuck in the queue while the remote agents are constantly dialed until
someone
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